[asterisk-users] Skype Messaging with Asterisk 10?
Hello, is every development on chan_skype out of the question after Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version that supports the new, shiny messaging-api in asterisk 10? Andreas... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype Messaging with Asterisk 10?
Hi Paul, is every development on chan_skype out of the question after Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version that supports the new, shiny messaging-api in asterisk 10? Nope, nobody submitted any patches for it. So anything now would have to be submitted into trunk, which would make Asterisk 11 the next version to support it. E, please correct me if i'm wrong, but the out-of-call-messaging-api is in asterisk 10 and currently supports sip and xmpp...? But i'm not asking for an extension of asterisk in any way, but of chan_skype that was sold 'til July... Again, assuming somebody submits a patch. Since when can someone submit a patch for chan_skype?? Did i miss an announcement that it has been opensourced? I'm under the impression that digium is the only party who *can* extend chan_skype... Kind regards, Andreas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Vocera Comm Badges
Hi, has someone ever got their hands on the Comm Badges from Vocera ( http://www.vocera.com/ ) and knows if they use anything standard and could work with asterisk, or does someone know an alternative to their really small, light devices? Regards, Andreas _ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Requirement for asterisk
i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. Can someone please throw that moron of the list?? _ Looking for a place to manage all your online stuff? Download the new Windows Live http://download.live.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDi broken in asterisk 1.4-svn?
Hi Guys, since about two weeks pbx_dundi.so from svn segfaults when i load it, 1.4.24 release works fine on the same box. Can someone tell me if that's something weird with my Fedora8 system or a possible bug in svn? Program terminated with signal 11, Segmentation fault. #0 0x in ?? () (gdb) bt #0 0x in ?? () #1 0x0881d00c in dundi_encrypt (trans=0x985e6e0, pack=0x985b5e0) at pbx_dundi.c:1298 #2 0x0881d607 in dundi_send (trans=0x985e6e0, cmdresp=10, flags=value optimized out, final=0, ied=0x1924210) at pbx_dundi.c:3081 #3 0x08824834 in do_register (data=0x985bf60) at pbx_dundi.c:4093 #4 0x080f0e31 in ast_sched_runq (con=0x9855780) at sched.c:363 #5 0x0881c89a in network_thread (ignore=0x0) at pbx_dundi.c:2164 #6 0x080fd7ab in dummy_start (data=0x9840ce0) at utils.c:856 #7 0x0013b50b in start_thread () from /lib/libpthread.so.0 #8 0x00263b2e in clone () from /lib/libc.so.6 Regards Andreas _ Find a way to cure that travel bug with MSN NZ Travel http://travel.msn.co.nz/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iChat voice (and maybe video?)
Hi Dudes, i searched for some time for an answer for this, i found some posting from John Todd half a decade ago [1], was there some chance in this? Is it somehow possible to voip from ichat to asterisk? If there's no light, is this something that could happen with enough founding, or is Mapple preventing this somehow (legal or technical)...? Regards, Andreas [1] http://lists.digium.com/pipermail/asterisk-dev/2003-July/001075.html _ Time for a change? SEEK and you shall find. http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fmsn%2Eseek%2Eco%2Enz%2FID%5FSEEKNZMAIN%5FUSR%2FPages%2Falliance%5Fhomepage%2Eascx%3FComeFrom%3Dmsnnz%26tracking%3Dsk%3Atl%3Asknz%3Amsnnz%3A0%3Ahottag%3Aflirt_t=757263783_r=SEEKNZ_tagline_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP voicemail import
Hi, i've switched from the old vm-storage to imap-storage. Is there a script that can import the old messages? Regards Andreas _ Buy, rent, invest property online today. http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fwww%2Eallrealestate%2Eco%2Enz%2Freview%2Fhome%2Dbuying%2Dinfo%2Ehtml%3Frsf%3Dmsnnz%5Ftextlink_t=26000_r=REA_NZ_tagline_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
Hi, Now the tough part...does anyone want to create an app to send notification to a cell phone to set/clear these bits? you can set these flags with a GSM-Phone connected to gnokii/gammu, or you can use a commercial SMS-Service like http://www.truesenses.com, both can be called via an externnotify script. On the downside, asterisk does not yet have a mechanism to track the status of an external MWI, so you have to do this yourself (send an SMS when you have new messages and clear the MWI after you've listened to the last one...) Regards, Andreas _ Find singles in your area with Match. http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fmatch%2Enz%2Emsn%2Ecom%2Fchannel%2Findex%2Easpx%3Ftrackingid%3D1043416_r=WL_EMAL_TAG_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softkeys wrong with chan_skinny
Hi, as noone out there seems to be able to maintain chan_sccp, i'm trying to switch to chan_skinny. With the newest 1.4 svn the Softkeys are mostly wrong/non functional. I see Redial NewCall CFwdAll more (more) CFwdBu... GPickUp Confrn more NewCall works, CFwdAll seems to toggle DnD, the rest of the buttons do notting. Any ideas how to fix this? Regards, Andreas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DUNDI behind NAT?
Hi, i'm having asterisk with sip working fine, including dundi lookups. The only problem i'm having is that the dundi answer allways contains my internal, private ip. Is there any way to set the targeting ip that is sent out in the dundi answer (to my public ip or any other where i want to receive the call)? Regards, Andreas. _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: nadi: branch 1.4 r61342 - /branches/1.4/channels/chan_misdn.c
Author: nadi Date: Wed Apr 11 05:52:28 2007 New Revision: 61342 URL: http://svn.digium.com/view/asterisk?view=revrev=61342 Log: AOCD's are now exported to asterisk channel variables. Modified: branches/1.4/channels/chan_misdn.c This is very cool, something i've waited for a long time :-). Is there a way to write this into CDR or sendtext() it to a channel? Regards, Andreas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AOCD - SendText()?
Hiya, i've just noticed that chan_misdn writes the AOCD information into a logfile. Has someone done a patch that sends this information via sendtext() to the active channel? At least some phones (like Cisco with chan_sccp and the snom-phones with SIP) can show this information on the display, what is kinda the idea with AOCD ;-) Regards, Andeas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AOCD - SendText()?
Hiya, i've just noticed that chan_misdn writes the AOCD information into a logfile. Has someone done a patch that sends this information via sendtext() to the active channel? At least some phones (like Cisco with chan_sccp and the snom-phones with SIP) can show this information on the display, what is kinda the idea with AOCD ;-) Regards, Andeas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AOCD - SendText()?
Hiya, i've just noticed that chan_misdn writes the AOCD information into a logfile. Has someone done a patch that sends this information via sendtext() to the active channel? At least some phones (like Cisco with chan_sccp and the snom-phones with SIP) can show this information on the display, what is kinda the idea with AOCD ;-) Regards, Andeas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicechanger update for asterisk 1.4
Hi, has someone here done a patch to use voicechanger with asterisk 1.4 and/or trunk? The Bug in bugs.digium.com was closed with the note that there is a version on http://www.lobstertech.com/code/voicechanger/ , but i cannot find something for 1.4 there... Greetings, Andreas _ Live Search delivers results the way you like it. Try live.com now! http://www.live.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to handle SIP-Callerid?
Hi, on ISDN there are the numbering plans that indicate if it's an national or an internation number. Is there something similar on SIP? How should i set a callerid to an internation number? complete e164, with, without an intl prefix (ie +, 011, 00 etc)...? How to a national number? Regards, Andreas. _ Discover fun and games at @ http://xtramsn.co.nz/kids ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AOC on misdn?
Hi, i can see AOC messages on the asterisk console. Can i sendtext() them to the caller or put them in cdr? Regards, Andreas. _ Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN crypto?
Hi, how do i have to specify the key's in misdn.conf? Does it even work in asterisk 1.2? When i try to do an encrypted call it get's rejected because i have 0 keys but specified key index 1?! Regards Andreas _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_skype license?
Hi guys, is there a comment from digium on the license of chan_skype? I could not find the GPL_KEY in the precompiled module, and they don't release the source. So i'm guessing, they'd need a commercial license...? Regards, Andreas _ Become a fitness fanatic @ http://xtramsn.co.nz/health ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Update for trunk?
Hi, someone out there has a patch for chan_sccp to work with trunk? Sergio seems to have abandoned the project and chan_skinny is still a long way from beeing really usable :-S Regards, Andreas. _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX/SIP to germany with own callerid?
Hi, does someone knows a IAX2 provider that allows to freely set the USER_NUMBER to the german pstn? It's no problem if setting the NETWORK_NUMBER is restricted, i'm using this for call-forwarding with the correct number, not something illegal... (i'm not sure about the terminology here, but i think USER_NUMBER is callerid and NETWORK_NUMBERis ANI in the US, correct?) regards, Andreas _ Read the latest Hollywood gossip @ http://xtramsn.co.nz/entertainment ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chinaroby VOIP phones? SECOND TIME!
Hi all, Do anyone have experience www.Chinaroby.com VOIP phones? Yes, we all have, that's why you ask for a SECOND TIME! I am very interested for models: PY-60 and PB-35 Phones. Good or bad experience with sip and IAX2, please comment. I did not find any comment on google try www.voip-info.org and lookup their history there. You don't want to do deals with a company who spam's a wiki _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Cisco 7960 Firmware 7.5
Hi, new features: RFC 3261 compliance (no TCP) RFC 3264 compliance RFC 3311 Compliance (display updates only, no media) Remote-Party-ID for display updatesA Remote-Party-ID header received in an INVITE or 200 OK will now update the display of the phone to accurately reflect the connected party New Configuration parameters sip_max_forwards and rfc_2543_hold REGISTER contact header sip.instance parameter support Does asterisk allready support supervised-transfers-with-correct-number (c sees number of a after b, who transferred a to c, hung up)...? Any other ideas what could be done with RFC3311/Remote-Party-ID-updates? Regards aa _ Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SecureTelephony
Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_darthvader.c?
Hi, on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks to some cool device from marshall :-) Is something like this possible with asterisk, or, asked a little more generic, can i somehow pipe an rtp-stream to an application via STDOUT and read it back via STDIN? Greetings, aa _ Read the latest Hollywood gossip @ http://xtramsn.co.nz/entertainment ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth with *
Does anyone know if one could use bluetooth on a cell phone with *? Would be nice to have your cell as an office phone combo. I heard that there is a bluetooth module for *? If so this should be possible? Yes, this is possible... NOTICE - This message contains privileged and confidential information intended only for the use of the addressee named above. If you are not the ...but, sadly, we can't tell you about it, as it is privileged and confidential information and we're not the addressee named above... SCNR _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: BRI in the US
Hi Brian, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? bye, aa _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_sip CallerPres support?
would it be hard to implement CallerPres support in chan_sip? There is support for outgoing calls, but this patch breakes incoming callerid: http://bugs.digium.com/bug_view_page.php?bug_id=0002471 Greez Andreas _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transparent SIP Server
Hi Guys, i need to do some kind of CDR for all clients inside my network, but they do not register/use the same sip-server, some of them use iptel, others fwd and various other services. Can i somehow put asterisk in the (control-)path between my clients and the other services (iptable-redirect like with a squid-proxy), so the clients don't have to change their settings and still register with their respective service, but asterisk does a complete CDR on every call? If thats not possible, anyone knows a software that supports this? SER? Regards, Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaphfc NT-mode can't dial outgoing
Hi Peter, As soon as i pick up the phone that is connected to the bri-card, asterisk jumps into extension s of the context that is specified in zapata.conf, if i have immediate=no then i hear the normal dialtone for about 1/10 of a second. I think you need to specify overlap dialing in the config file. No, i've already tried that. My zapata.conf looks like this: [channels] language=de switchtype=euroisdn signalling=bri_net_ptmp pridialplan=unknown prilocaldialplan=unknown pritrustusercid=no usecallingpres=yes echocancel=yes immediate=no group=1 context=default channel = 1-2 overlapdial=yes Thanks, Andreas _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zaphfc NT-mode can't dial outgoing
Hi Tim, As soon as i pick up the phone that is connected to the bri-card, asterisk jumps into extension s of the context that is specified in The 'channel' line has to be the last line of the declaration. Try moving the 'overlap dial' line up above the 'channel' line. Doh. That fixed the problem, thanks a lot :-) Regards, Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 2610XM and Asterisk
Hi, Is the VIC-2BRI compatible with the 2610XM? What IOS needs to be loaded? I'm not sure about the 2610XM, but the V1 with a VIC-2BRI work's fine in a C3620 with a PLUS image, c3620-is-mz.122-15.T1.bin Greetings, Andreas _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaphfc NT-mode can't dial outgoing
Hi Guys, i've successfully installed the current bristuff, everything works fine, exept one thing: As soon as i pick up the phone that is connected to the bri-card, asterisk jumps into extension s of the context that is specified in zapata.conf, if i have immediate=no then i hear the normal dialtone for about 1/10 of a second. Any hints what i'm doing wrong? If i dial the number *before* i pick up the phone everything works as expected. Regards, Andreas. _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Earthlink Releases SIP Based P2P File-Sharing App
This is BAAAD! Now even SIP get's tainted... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IBM to Open Voice Recognition Software
http://developers.slashdot.org/developers/04/09/13/1058241.shtml _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Really long first ring, then normal
Hiya, I've been seeing this lately with our Asterisk PBX as well. With Asterisk CVS HEAD 05-21, the ringing was normal. After our last upgrade, CVS HEAD 07-03, the first ring is almost always one continuous ring that lasts about 2 to 3 seconds. We're noticing other problems too, such as hangup detection. This worked flawlessly for us with CVS HEAD 05-21 (we have a TDM400P card with 1 FXS and 1 FXO). But, now Asterisk isn't detecting hangups at all. I'm not sure at the moment if something on the actual POTS line has changed, or if it's a problem with the CVS version we're running. Anyone else noticing strange behaviour such as the above? jup, me2: If i'm calling my Cisco 7960 from my cellphone, the Cisco rings 3-5 times before i hear the ring on the cellphones. I tried progress=yes and no in sip.conf, no difference. The calls come in on a BRI with chan_capi, OR on a Cisco 3620 with VIC-2BRI, it's the same problem both ways, so its neither capi nor a SIP problem... I'm using CVS from today. I've just opened a bug with ID2062, maybe someone knows when exacty this problem started... Andreas. _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Audio filters (was: feature - VM gain adjust?)
Hiya, This is an excellent idea, and is extendable outside of the narrow scope of audio quality improvement. I was playing with this concept a while back, and trying to find programmers for a few ideas I have. I'll air them here, so I can take some credit for being the first clever monkey to publicly talk about integration into Asterisk (or any other VoIP system, as far as I know): - voice disguise/modulation. Think about how many customers you'd get with a module that sounds like they're Mickey Mouse. You think: 'That's really stupid!' but then look at how many Please press 7 for Darth Vader -- I am your father, Luke :-D - voice stress analysis. If you're dumping the audio through a filter, there's no reason you can't simply extract data from it instead of alter the audio path. A one-way background audio carrier tone to the listener might change pitch during stress events. Is there allready some application to do a voice stress analysis? I guess developing something like this from scratch would be very hard... - customized background noise. This is apparently already the rage in Asia somewhere with some cell phone carriers - insertion of background sounds customized to the user's tastes (forest, construction site, bar, office environment, airport, etc.) which can be used for either pleasant diversion or for disguise of location. yeah, this would rock. Honey, i've to stop talkin', the Dentist want's to start drilling. For some Cellphones, this allready exists: http://www.simeda.com/soundercover.html This could also be used to do (MusicDuringCall. Get a call from the army and you play Status Quo (http://www.france-jeunes.net/paroles/index.php?tid=MTkwOTQ=) :-) I have a few more, even, but as is typical, these will remain on the drawing board until someone coughs up some dough to make them happen. No time, no time, no time... Hey, no normal person uses asterisk at home anyway, so there HAS to be some geek out there who also wants this AND can code :-D Bye Andreas _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Remote-Party-ID / Bug #2012
Hello Guys, after an update to cvs head (thanks oej!) my CiscoGW can now flag unkown caller's to Number AND Name Unkown. Before i again open a new bug (which isn't a bug :-)), can someone confirm this: - PrivacyManager does not recognize this as an unknown number - it's not possible to set ANY CID with SetCallerID, it allways stays on Unknown (with chan_capi i had to do a SetCallerID() to get PM to recognize it...) - is there a variable with the stat of the privacy indicator in the remote-party-id? - Is there a way to set CALLERIDNUM to an alphanumeric value? I've seen this with asterisk, anonymous and unknown, so it is possible with the Cisco 7960... Thanks and regards, Andreas _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Chan_Capi Down
Hiya, Looks like this may need a bug report? We are all getting the same errors. Sure, but i guess a bug to bugs.digium.com will be rejected, chan_capi is not in CVS. Maybe [EMAIL PROTECTED] could do a fix, :-D PLEEASE :-D? BTW: kapejod, any chances to disclaim chan_capi to digium? It would safe some troubles if it was in CVS... Outgoing is fine for me. yes, no problem with outgoing. bye. aa _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Chan_Capi Down
Same here :-( asterisk show's this error in the same moment i'm trying to pick up an incoming call: Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 This problem starts with cvs update -D 6/21/04 21:00:00 CET If i revert back to cvs update -D 6/21/04 18:00:00 CET the problem is gone. -- original message -- I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP response 488 to special ext/pri?
Hi all, if you turn on Anonymous Call Block on a Cisco 7960, the phone rejects incoming calls that have Anonymous as callerid with -- Got SIP response 488 Not Acceptable Here back from 192.168.1.1 == No one is available to answer at this time Normaly, the next priority after the dial-command is executed. Is there any way to send this to a special extension or priority, ie a or +51? Regards, aa _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 Beta Codec
Hi Guys, - if i buy the codec online NOW, will this license work with the new one, or shoud i wait 'til it is no longer beta? - how does the registration work? Will there be a key-file that i can backup and use on a new installation of the same machine with the same hardware, or do i have to re-register on any new installation (and therefore i'm screwed after the second new installation)? If there's no key file, will it at least survive an image-backup (ghost) of the machine? Regards, Andreas _ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use buttons (other than #) after call is bridged?
Hi, can i somehow use the other buttons to execute some apps, *without* hanging up the call? Something like: exten = s,1,Dial/SIP(1234)|4,5,7,9 exten = 4,1,Monitor(wav) exten = 5,1,SIPDtmfMode(inband) exten = 7,1,AGI(turnoncoffeemachine.agi) exten = 9,1,System(smbnuke boss) Regards, AA _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi
Hi Guys, does anyone know how to fix chan_capi to work with the current CVS HEAD? It's no longer possible to compile after the recent changes in the locking... Regards, Andreas _ Theres never been a better time to get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk + Fritz!PCI + CAPI
Hello,
The ONLY issue I have is that I don't get ringing dialback so
calling out gives a silence until the other party picks up
Have you turned on early B3?
S,1,Dial,CAPI/12345678:b${EXTEN}|30 (always early B3)
(plus the recent changes to locks in * required a tweak to the
chan_capi source to match).
could you post this tweak here? I'm stuck with CVS-STABLE ;-(
Regards,
Andreas
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[Asterisk-Users] (no subject)
Dialing in from the pstn to sip phones (x-lite softphone on winders and a grandstream handytone-286 ata), I hear the sip phone ring a few times, I ran into the same thing with Cisco 7960. Looks like the logic in the sip channel has changed recently. Add a ,r to the end of your Dial statements in extensions.conf and the issue should go away. Does anyone know if this was done intentionally? I don't want to open a bug for something that's really a feature, but i simply can't think of any reason someone want's to update their whole extensions.conf. Can someone tell me what i have to change in the source to get the old (correct :-) way ...? Regards, Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] It's dead jim!
Hi Guys, with cvs from today, incoming calls via capi stoped working. the call comes in (visible in capi debug), but asterisk just ignores it; to the caller it's like theres no phone connected... am i the only one with this problem...? CVS from yesterday moring workes fine... Greez Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco LDAP directory
Hi, does anyone has an LDAP based directory for the Cisco 7900 Series? I found some directorys based on sql, but an ldap directory would allow syncronisation with evolution, mozilla, multisync (Palm, OPIE, Mobile phones etc...) Greez Andreas _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)
Hiya, Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly straightforward. The release notes indicate that you can trigger other ringtones on the phone (in the section Support for SIP Alert-Info Header), but I can't get anywhere with it. the only thing i'm getting, is using exten = 555,1,SetVar(ALERT_INFO=7) to get the phone to ring Ring Ring... Ring Ring with the preset Ring Type. It doesn't matter what alert_info i set, no matter if i use Bellcore-dr1 or anything else. What are you using, and how does the phone react to it...? regards, aa _ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN Personals! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
Hi Brian, Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. is cepstral a special tts-api, or does this mean, we can use every windows(tm) tts-engine on the market...? Even ATT Natural Voices...? Can we allready test this app, or is this a closed source thingy...? Greez Andreas _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7960 MGCP dialtone problems, part 1 [long]
Hiya, I've been trying on and off again for several months to get my 7960 (MGCP 5.3) working with * with no success. As you know, working MGCP configs for non-ATA Ciscos seem to be very hard to come by. I'm not shooting for the moon here, just trying to get dialtone at the moment. The problem I'd like to focus on today: I only get dialtone when I go off-hook (via the Speaker button, if it matters) maybe once every 3 tries. If it fails, or after I've successfully gotten the dialtone once, the phone will not get it again until it has been power-cycled. I could not get 5.3 to work, but 6.1 seems to work. Basic Phone that is, i don't get *any* buttons on the phone, i guess this is a problem with CARD.XML, the only version on CCO ist for version 3.0 (!). If anyone has a working one, please post it on the list ;-) Regards, aa _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID Presentment on PRI...
Hiya,
Is anybody out there currently able to set CIDName to be something
different than the reverse lookup name? My goal is not to spoof the
White House, btw, but it makes a fun example.
Beware the three-letter agencies. Beware even more the two-letter ones.
yeah, impersonating the White House sounds like asking for trouble ;-]
On topic: on most networks in europe you can't even fake the number,
everything that does not belong to your BRI/PRI is simply rewritten to
the primary number. BUT there is a service called UUS1 (UserUserSignaling),
just some text which can sent with the setup-message. A few phones (like
Ascom or Tiptel)
can set and display this message.
@kapejod, is there a change for a chan_capi that can read/set the uus1
to/from
the ${CALLERIDNAME}...? A lot of (really big) pbxes (Meridian etc) also set
the name
of the caller...
Regards,
andreas
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[Asterisk-Users] Client for P800/P900
Hi Guys, is there a client which can be used on the SonyEricsson P800/P900...? IAX would be cool, but i take anything that can connect (via bluetooth) to an asterisk-server ;-). The phone is Symbian, and can also execute java-stuff... Greez Andreas _ Find your perfect match @ http://personals.xtramsn.co.nz with XtraMSN Personals! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TRIP / RFC2871
Hi guys, is there support for TRIP (Telephone Routing over IP, TFC2871) within asterisk? There is an implementation from Vovida[1], would it be possible to use this with asterisk? bye... [1] http://www.vovida.org/protocols/downloads/trip/ _ MSN 8 helps eliminate e-mail viruses. Get 2 months FREE*. http://join.msn.com/?page=features/virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
